include/SDL_audio.h
author Ryan C. Gordon <icculus@icculus.org>
Mon, 27 Feb 2017 10:11:40 -0500
changeset 10920 ffdf6433884c
parent 10845 c2b0d3d180f4
child 11096 819384789a7a
permissions -rw-r--r--
audio: clarified what SDL_AudioSpec::samples is, removed note about power of 2.

These don't have to be power-of-2 sizes anymore because of SDL_AudioStream,
and the new resampler, but also, many platforms don't give you power-of-2 DMA
buffer in the first place!
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/*
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  Simple DirectMedia Layer
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  Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
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  This software is provided 'as-is', without any express or implied
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  warranty.  In no event will the authors be held liable for any damages
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  arising from the use of this software.
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  Permission is granted to anyone to use this software for any purpose,
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  including commercial applications, and to alter it and redistribute it
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  freely, subject to the following restrictions:
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  1. The origin of this software must not be misrepresented; you must not
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     claim that you wrote the original software. If you use this software
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     in a product, an acknowledgment in the product documentation would be
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     appreciated but is not required.
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  2. Altered source versions must be plainly marked as such, and must not be
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     misrepresented as being the original software.
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  3. This notice may not be removed or altered from any source distribution.
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*/
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/**
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 *  \file SDL_audio.h
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 *
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 *  Access to the raw audio mixing buffer for the SDL library.
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 */
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#ifndef SDL_audio_h_
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#define SDL_audio_h_
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#include "SDL_stdinc.h"
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#include "SDL_error.h"
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#include "SDL_endian.h"
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#include "SDL_mutex.h"
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#include "SDL_thread.h"
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#include "SDL_rwops.h"
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#include "begin_code.h"
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/* Set up for C function definitions, even when using C++ */
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#ifdef __cplusplus
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extern "C" {
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#endif
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/**
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 *  \brief Audio format flags.
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 *
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 *  These are what the 16 bits in SDL_AudioFormat currently mean...
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 *  (Unspecified bits are always zero).
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 *
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 *  \verbatim
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    ++-----------------------sample is signed if set
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    ||
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    ||       ++-----------sample is bigendian if set
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    ||       ||
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    ||       ||          ++---sample is float if set
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    ||       ||          ||
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    ||       ||          || +---sample bit size---+
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    ||       ||          || |                     |
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    15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
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    \endverbatim
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 *
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 *  There are macros in SDL 2.0 and later to query these bits.
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 */
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typedef Uint16 SDL_AudioFormat;
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/**
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 *  \name Audio flags
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 */
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/* @{ */
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#define SDL_AUDIO_MASK_BITSIZE       (0xFF)
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#define SDL_AUDIO_MASK_DATATYPE      (1<<8)
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#define SDL_AUDIO_MASK_ENDIAN        (1<<12)
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#define SDL_AUDIO_MASK_SIGNED        (1<<15)
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#define SDL_AUDIO_BITSIZE(x)         (x & SDL_AUDIO_MASK_BITSIZE)
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#define SDL_AUDIO_ISFLOAT(x)         (x & SDL_AUDIO_MASK_DATATYPE)
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#define SDL_AUDIO_ISBIGENDIAN(x)     (x & SDL_AUDIO_MASK_ENDIAN)
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#define SDL_AUDIO_ISSIGNED(x)        (x & SDL_AUDIO_MASK_SIGNED)
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#define SDL_AUDIO_ISINT(x)           (!SDL_AUDIO_ISFLOAT(x))
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#define SDL_AUDIO_ISLITTLEENDIAN(x)  (!SDL_AUDIO_ISBIGENDIAN(x))
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#define SDL_AUDIO_ISUNSIGNED(x)      (!SDL_AUDIO_ISSIGNED(x))
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/**
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 *  \name Audio format flags
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 *
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 *  Defaults to LSB byte order.
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 */
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/* @{ */
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#define AUDIO_U8        0x0008  /**< Unsigned 8-bit samples */
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#define AUDIO_S8        0x8008  /**< Signed 8-bit samples */
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#define AUDIO_U16LSB    0x0010  /**< Unsigned 16-bit samples */
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#define AUDIO_S16LSB    0x8010  /**< Signed 16-bit samples */
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#define AUDIO_U16MSB    0x1010  /**< As above, but big-endian byte order */
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#define AUDIO_S16MSB    0x9010  /**< As above, but big-endian byte order */
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#define AUDIO_U16       AUDIO_U16LSB
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#define AUDIO_S16       AUDIO_S16LSB
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/* @} */
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/**
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 *  \name int32 support
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 */
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/* @{ */
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#define AUDIO_S32LSB    0x8020  /**< 32-bit integer samples */
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#define AUDIO_S32MSB    0x9020  /**< As above, but big-endian byte order */
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#define AUDIO_S32       AUDIO_S32LSB
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/* @} */
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/**
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 *  \name float32 support
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 */
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/* @{ */
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#define AUDIO_F32LSB    0x8120  /**< 32-bit floating point samples */
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#define AUDIO_F32MSB    0x9120  /**< As above, but big-endian byte order */
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#define AUDIO_F32       AUDIO_F32LSB
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/* @} */
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/**
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 *  \name Native audio byte ordering
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 */
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/* @{ */
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#if SDL_BYTEORDER == SDL_LIL_ENDIAN
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#define AUDIO_U16SYS    AUDIO_U16LSB
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#define AUDIO_S16SYS    AUDIO_S16LSB
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#define AUDIO_S32SYS    AUDIO_S32LSB
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#define AUDIO_F32SYS    AUDIO_F32LSB
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#else
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#define AUDIO_U16SYS    AUDIO_U16MSB
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#define AUDIO_S16SYS    AUDIO_S16MSB
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#define AUDIO_S32SYS    AUDIO_S32MSB
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#define AUDIO_F32SYS    AUDIO_F32MSB
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#endif
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/* @} */
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/**
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 *  \name Allow change flags
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 *
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 *  Which audio format changes are allowed when opening a device.
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 */
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/* @{ */
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#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE    0x00000001
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#define SDL_AUDIO_ALLOW_FORMAT_CHANGE       0x00000002
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#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE     0x00000004
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#define SDL_AUDIO_ALLOW_ANY_CHANGE          (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE)
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/* @} */
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/* @} *//* Audio flags */
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/**
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 *  This function is called when the audio device needs more data.
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 *
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 *  \param userdata An application-specific parameter saved in
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 *                  the SDL_AudioSpec structure
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 *  \param stream A pointer to the audio data buffer.
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 *  \param len    The length of that buffer in bytes.
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 *
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 *  Once the callback returns, the buffer will no longer be valid.
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 *  Stereo samples are stored in a LRLRLR ordering.
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 *
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 *  You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
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 *  you like. Just open your audio device with a NULL callback.
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 */
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typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
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                                            int len);
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/**
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 *  The calculated values in this structure are calculated by SDL_OpenAudio().
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 */
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typedef struct SDL_AudioSpec
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{
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    int freq;                   /**< DSP frequency -- samples per second */
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    SDL_AudioFormat format;     /**< Audio data format */
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    Uint8 channels;             /**< Number of channels: 1 mono, 2 stereo */
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    Uint8 silence;              /**< Audio buffer silence value (calculated) */
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    Uint16 samples;             /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */
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    Uint16 padding;             /**< Necessary for some compile environments */
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    Uint32 size;                /**< Audio buffer size in bytes (calculated) */
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    SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
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    void *userdata;             /**< Userdata passed to callback (ignored for NULL callbacks). */
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} SDL_AudioSpec;
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struct SDL_AudioCVT;
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typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
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                                          SDL_AudioFormat format);
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/**
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 *  A structure to hold a set of audio conversion filters and buffers.
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 *
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 *  Note that various parts of the conversion pipeline can take advantage
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 *  of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
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 *  you to pass it aligned data, but can possibly run much faster if you
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 *  set both its (buf) field to a pointer that is aligned to 16 bytes, and its
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 *  (len) field to something that's a multiple of 16, if possible.
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 */
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#ifdef __GNUC__
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/* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
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   pad it out to 88 bytes to guarantee ABI compatibility between compilers.
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   vvv
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   The next time we rev the ABI, make sure to size the ints and add padding.
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*/
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#define SDL_AUDIOCVT_PACKED __attribute__((packed))
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#else
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#define SDL_AUDIOCVT_PACKED
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#endif
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/* */
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typedef struct SDL_AudioCVT
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{
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    int needed;                 /**< Set to 1 if conversion possible */
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    SDL_AudioFormat src_format; /**< Source audio format */
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    SDL_AudioFormat dst_format; /**< Target audio format */
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    double rate_incr;           /**< Rate conversion increment */
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    Uint8 *buf;                 /**< Buffer to hold entire audio data */
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    int len;                    /**< Length of original audio buffer */
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    int len_cvt;                /**< Length of converted audio buffer */
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    int len_mult;               /**< buffer must be len*len_mult big */
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    double len_ratio;           /**< Given len, final size is len*len_ratio */
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    SDL_AudioFilter filters[10];        /**< Filter list */
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    int filter_index;           /**< Current audio conversion function */
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} SDL_AUDIOCVT_PACKED SDL_AudioCVT;
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/* Function prototypes */
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/**
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 *  \name Driver discovery functions
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 *
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 *  These functions return the list of built in audio drivers, in the
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 *  order that they are normally initialized by default.
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 */
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/* @{ */
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extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
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extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
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/* @} */
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/**
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 *  \name Initialization and cleanup
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 *
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 *  \internal These functions are used internally, and should not be used unless
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 *            you have a specific need to specify the audio driver you want to
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 *            use.  You should normally use SDL_Init() or SDL_InitSubSystem().
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 */
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/* @{ */
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extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
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extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
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/* @} */
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/**
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 *  This function returns the name of the current audio driver, or NULL
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 *  if no driver has been initialized.
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 */
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extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
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/**
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 *  This function opens the audio device with the desired parameters, and
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 *  returns 0 if successful, placing the actual hardware parameters in the
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 *  structure pointed to by \c obtained.  If \c obtained is NULL, the audio
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 *  data passed to the callback function will be guaranteed to be in the
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 *  requested format, and will be automatically converted to the hardware
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 *  audio format if necessary.  This function returns -1 if it failed
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 *  to open the audio device, or couldn't set up the audio thread.
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 *
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 *  When filling in the desired audio spec structure,
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 *    - \c desired->freq should be the desired audio frequency in samples-per-
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 *      second.
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 *    - \c desired->format should be the desired audio format.
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 *    - \c desired->samples is the desired size of the audio buffer, in
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 *      samples.  This number should be a power of two, and may be adjusted by
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 *      the audio driver to a value more suitable for the hardware.  Good values
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 *      seem to range between 512 and 8096 inclusive, depending on the
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 *      application and CPU speed.  Smaller values yield faster response time,
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 *      but can lead to underflow if the application is doing heavy processing
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 *      and cannot fill the audio buffer in time.  A stereo sample consists of
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 *      both right and left channels in LR ordering.
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 *      Note that the number of samples is directly related to time by the
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 *      following formula:  \code ms = (samples*1000)/freq \endcode
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 *    - \c desired->size is the size in bytes of the audio buffer, and is
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 *      calculated by SDL_OpenAudio().
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 *    - \c desired->silence is the value used to set the buffer to silence,
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 *      and is calculated by SDL_OpenAudio().
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 *    - \c desired->callback should be set to a function that will be called
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 *      when the audio device is ready for more data.  It is passed a pointer
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 *      to the audio buffer, and the length in bytes of the audio buffer.
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 *      This function usually runs in a separate thread, and so you should
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 *      protect data structures that it accesses by calling SDL_LockAudio()
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 *      and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL
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 *      pointer here, and call SDL_QueueAudio() with some frequency, to queue
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 *      more audio samples to be played (or for capture devices, call
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 *      SDL_DequeueAudio() with some frequency, to obtain audio samples).
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 *    - \c desired->userdata is passed as the first parameter to your callback
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 *      function. If you passed a NULL callback, this value is ignored.
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 *
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 *  The audio device starts out playing silence when it's opened, and should
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 *  be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready
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 *  for your audio callback function to be called.  Since the audio driver
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 *  may modify the requested size of the audio buffer, you should allocate
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 *  any local mixing buffers after you open the audio device.
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 */
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extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
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                                          SDL_AudioSpec * obtained);
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/**
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 *  SDL Audio Device IDs.
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 *
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 *  A successful call to SDL_OpenAudio() is always device id 1, and legacy
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 *  SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
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 *  always returns devices >= 2 on success. The legacy calls are good both
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 *  for backwards compatibility and when you don't care about multiple,
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 *  specific, or capture devices.
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 */
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typedef Uint32 SDL_AudioDeviceID;
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/**
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 *  Get the number of available devices exposed by the current driver.
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 *  Only valid after a successfully initializing the audio subsystem.
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 *  Returns -1 if an explicit list of devices can't be determined; this is
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 *  not an error. For example, if SDL is set up to talk to a remote audio
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 *  server, it can't list every one available on the Internet, but it will
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 *  still allow a specific host to be specified to SDL_OpenAudioDevice().
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 *
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 *  In many common cases, when this function returns a value <= 0, it can still
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 *  successfully open the default device (NULL for first argument of
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 *  SDL_OpenAudioDevice()).
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 */
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extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
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/**
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 *  Get the human-readable name of a specific audio device.
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 *  Must be a value between 0 and (number of audio devices-1).
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 *  Only valid after a successfully initializing the audio subsystem.
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 *  The values returned by this function reflect the latest call to
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 *  SDL_GetNumAudioDevices(); recall that function to redetect available
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 *  hardware.
slouken@7191
   333
 *
slouken@3407
   334
 *  The string returned by this function is UTF-8 encoded, read-only, and
icculus@2049
   335
 *  managed internally. You are not to free it. If you need to keep the
icculus@2049
   336
 *  string for any length of time, you should make your own copy of it, as it
icculus@2049
   337
 *  will be invalid next time any of several other SDL functions is called.
icculus@1964
   338
 */
icculus@2049
   339
extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
icculus@2049
   340
                                                           int iscapture);
icculus@1964
   341
icculus@1964
   342
slouken@3407
   343
/**
slouken@3407
   344
 *  Open a specific audio device. Passing in a device name of NULL requests
icculus@2049
   345
 *  the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
slouken@7191
   346
 *
slouken@3407
   347
 *  The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
icculus@2049
   348
 *  some drivers allow arbitrary and driver-specific strings, such as a
icculus@2049
   349
 *  hostname/IP address for a remote audio server, or a filename in the
icculus@2049
   350
 *  diskaudio driver.
slouken@7191
   351
 *
slouken@3407
   352
 *  \return 0 on error, a valid device ID that is >= 2 on success.
slouken@7191
   353
 *
icculus@2049
   354
 *  SDL_OpenAudio(), unlike this function, always acts on device ID 1.
icculus@1964
   355
 */
slouken@1967
   356
extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
slouken@1967
   357
                                                              *device,
slouken@1967
   358
                                                              int iscapture,
slouken@1967
   359
                                                              const
slouken@1967
   360
                                                              SDL_AudioSpec *
slouken@1967
   361
                                                              desired,
slouken@1967
   362
                                                              SDL_AudioSpec *
slouken@2866
   363
                                                              obtained,
slouken@2866
   364
                                                              int
slouken@2866
   365
                                                              allowed_changes);
icculus@1964
   366
icculus@1964
   367
icculus@1964
   368
slouken@3407
   369
/**
slouken@3407
   370
 *  \name Audio state
slouken@7191
   371
 *
slouken@3407
   372
 *  Get the current audio state.
slouken@0
   373
 */
gabomdq@7678
   374
/* @{ */
slouken@1895
   375
typedef enum
slouken@1895
   376
{
slouken@1895
   377
    SDL_AUDIO_STOPPED = 0,
slouken@1895
   378
    SDL_AUDIO_PLAYING,
slouken@1895
   379
    SDL_AUDIO_PAUSED
slouken@3537
   380
} SDL_AudioStatus;
slouken@3537
   381
extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
slouken@0
   382
slouken@3537
   383
extern DECLSPEC SDL_AudioStatus SDLCALL
slouken@1967
   384
SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
gabomdq@7678
   385
/* @} *//* Audio State */
icculus@1964
   386
slouken@3407
   387
/**
slouken@3407
   388
 *  \name Pause audio functions
slouken@7191
   389
 *
slouken@3407
   390
 *  These functions pause and unpause the audio callback processing.
slouken@3407
   391
 *  They should be called with a parameter of 0 after opening the audio
slouken@3407
   392
 *  device to start playing sound.  This is so you can safely initialize
slouken@3407
   393
 *  data for your callback function after opening the audio device.
slouken@3407
   394
 *  Silence will be written to the audio device during the pause.
slouken@0
   395
 */
gabomdq@7678
   396
/* @{ */
slouken@337
   397
extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
icculus@1964
   398
extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
icculus@1964
   399
                                                  int pause_on);
gabomdq@7678
   400
/* @} *//* Pause audio functions */
slouken@0
   401
slouken@3407
   402
/**
slouken@3407
   403
 *  This function loads a WAVE from the data source, automatically freeing
slouken@3407
   404
 *  that source if \c freesrc is non-zero.  For example, to load a WAVE file,
slouken@3407
   405
 *  you could do:
slouken@3407
   406
 *  \code
slouken@7191
   407
 *      SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
slouken@3407
   408
 *  \endcode
slouken@0
   409
 *
slouken@3407
   410
 *  If this function succeeds, it returns the given SDL_AudioSpec,
slouken@3407
   411
 *  filled with the audio data format of the wave data, and sets
slouken@3407
   412
 *  \c *audio_buf to a malloc()'d buffer containing the audio data,
slouken@3407
   413
 *  and sets \c *audio_len to the length of that audio buffer, in bytes.
slouken@7191
   414
 *  You need to free the audio buffer with SDL_FreeWAV() when you are
slouken@3407
   415
 *  done with it.
slouken@0
   416
 *
slouken@7191
   417
 *  This function returns NULL and sets the SDL error message if the
slouken@7191
   418
 *  wave file cannot be opened, uses an unknown data format, or is
slouken@3407
   419
 *  corrupt.  Currently raw and MS-ADPCM WAVE files are supported.
slouken@0
   420
 */
slouken@1895
   421
extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
slouken@1895
   422
                                                      int freesrc,
slouken@1895
   423
                                                      SDL_AudioSpec * spec,
slouken@1895
   424
                                                      Uint8 ** audio_buf,
slouken@1895
   425
                                                      Uint32 * audio_len);
slouken@0
   426
slouken@7191
   427
/**
slouken@3407
   428
 *  Loads a WAV from a file.
slouken@3407
   429
 *  Compatibility convenience function.
slouken@3407
   430
 */
slouken@0
   431
#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
slouken@7191
   432
    SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
slouken@0
   433
slouken@3407
   434
/**
slouken@3407
   435
 *  This function frees data previously allocated with SDL_LoadWAV_RW()
slouken@0
   436
 */
slouken@1895
   437
extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
slouken@0
   438
slouken@3407
   439
/**
slouken@3407
   440
 *  This function takes a source format and rate and a destination format
slouken@3407
   441
 *  and rate, and initializes the \c cvt structure with information needed
slouken@3407
   442
 *  by SDL_ConvertAudio() to convert a buffer of audio data from one format
slouken@3407
   443
 *  to the other.
slouken@7191
   444
 *
slouken@3407
   445
 *  \return -1 if the format conversion is not supported, 0 if there's
icculus@1964
   446
 *  no conversion needed, or 1 if the audio filter is set up.
slouken@0
   447
 */
slouken@1895
   448
extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
icculus@1983
   449
                                              SDL_AudioFormat src_format,
slouken@1895
   450
                                              Uint8 src_channels,
slouken@1895
   451
                                              int src_rate,
icculus@1983
   452
                                              SDL_AudioFormat dst_format,
slouken@1895
   453
                                              Uint8 dst_channels,
slouken@1895
   454
                                              int dst_rate);
slouken@0
   455
slouken@3407
   456
/**
slouken@3407
   457
 *  Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
slouken@3407
   458
 *  created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of
slouken@3407
   459
 *  audio data in the source format, this function will convert it in-place
slouken@3407
   460
 *  to the desired format.
slouken@7191
   461
 *
slouken@3407
   462
 *  The data conversion may expand the size of the audio data, so the buffer
slouken@3407
   463
 *  \c cvt->buf should be allocated after the \c cvt structure is initialized by
slouken@3407
   464
 *  SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
slouken@0
   465
 */
slouken@1895
   466
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
slouken@0
   467
slouken@3407
   468
#define SDL_MIX_MAXVOLUME 128
slouken@3407
   469
/**
slouken@3407
   470
 *  This takes two audio buffers of the playing audio format and mixes
slouken@3407
   471
 *  them, performing addition, volume adjustment, and overflow clipping.
slouken@3407
   472
 *  The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME
slouken@3407
   473
 *  for full audio volume.  Note this does not change hardware volume.
slouken@3407
   474
 *  This is provided for convenience -- you can mix your own audio data.
slouken@0
   475
 */
slouken@1895
   476
extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
slouken@1895
   477
                                          Uint32 len, int volume);
slouken@0
   478
slouken@3407
   479
/**
slouken@3407
   480
 *  This works like SDL_MixAudio(), but you specify the audio format instead of
icculus@1964
   481
 *  using the format of audio device 1. Thus it can be used when no audio
icculus@1964
   482
 *  device is open at all.
icculus@1964
   483
 */
slouken@1967
   484
extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
slouken@1967
   485
                                                const Uint8 * src,
icculus@1964
   486
                                                SDL_AudioFormat format,
icculus@1964
   487
                                                Uint32 len, int volume);
icculus@1964
   488
slouken@3407
   489
/**
icculus@9012
   490
 *  Queue more audio on non-callback devices.
icculus@9012
   491
 *
icculus@10262
   492
 *  (If you are looking to retrieve queued audio from a non-callback capture
icculus@10262
   493
 *  device, you want SDL_DequeueAudio() instead. This will return -1 to
icculus@10262
   494
 *  signify an error if you use it with capture devices.)
icculus@10262
   495
 *
icculus@9012
   496
 *  SDL offers two ways to feed audio to the device: you can either supply a
icculus@9012
   497
 *  callback that SDL triggers with some frequency to obtain more audio
icculus@9012
   498
 *  (pull method), or you can supply no callback, and then SDL will expect
icculus@9012
   499
 *  you to supply data at regular intervals (push method) with this function.
icculus@9012
   500
 *
icculus@9012
   501
 *  There are no limits on the amount of data you can queue, short of
icculus@9012
   502
 *  exhaustion of address space. Queued data will drain to the device as
icculus@9012
   503
 *  necessary without further intervention from you. If the device needs
icculus@9012
   504
 *  audio but there is not enough queued, it will play silence to make up
icculus@9012
   505
 *  the difference. This means you will have skips in your audio playback
icculus@9012
   506
 *  if you aren't routinely queueing sufficient data.
icculus@9012
   507
 *
icculus@9012
   508
 *  This function copies the supplied data, so you are safe to free it when
icculus@9012
   509
 *  the function returns. This function is thread-safe, but queueing to the
icculus@9012
   510
 *  same device from two threads at once does not promise which buffer will
icculus@9012
   511
 *  be queued first.
icculus@9012
   512
 *
icculus@9012
   513
 *  You may not queue audio on a device that is using an application-supplied
icculus@9012
   514
 *  callback; doing so returns an error. You have to use the audio callback
icculus@9012
   515
 *  or queue audio with this function, but not both.
icculus@9012
   516
 *
icculus@9012
   517
 *  You should not call SDL_LockAudio() on the device before queueing; SDL
icculus@9012
   518
 *  handles locking internally for this function.
icculus@9012
   519
 *
icculus@9012
   520
 *  \param dev The device ID to which we will queue audio.
icculus@9012
   521
 *  \param data The data to queue to the device for later playback.
icculus@9012
   522
 *  \param len The number of bytes (not samples!) to which (data) points.
icculus@9012
   523
 *  \return zero on success, -1 on error.
icculus@9012
   524
 *
icculus@9012
   525
 *  \sa SDL_GetQueuedAudioSize
icculus@9012
   526
 *  \sa SDL_ClearQueuedAudio
icculus@9012
   527
 */
icculus@9012
   528
extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
icculus@9012
   529
icculus@9012
   530
/**
icculus@10262
   531
 *  Dequeue more audio on non-callback devices.
icculus@10262
   532
 *
icculus@10262
   533
 *  (If you are looking to queue audio for output on a non-callback playback
icculus@10262
   534
 *  device, you want SDL_QueueAudio() instead. This will always return 0
icculus@10262
   535
 *  if you use it with playback devices.)
icculus@10262
   536
 *
icculus@10262
   537
 *  SDL offers two ways to retrieve audio from a capture device: you can
icculus@10262
   538
 *  either supply a callback that SDL triggers with some frequency as the
icculus@10262
   539
 *  device records more audio data, (push method), or you can supply no
icculus@10262
   540
 *  callback, and then SDL will expect you to retrieve data at regular
icculus@10262
   541
 *  intervals (pull method) with this function.
icculus@10262
   542
 *
icculus@10262
   543
 *  There are no limits on the amount of data you can queue, short of
icculus@10262
   544
 *  exhaustion of address space. Data from the device will keep queuing as
icculus@10262
   545
 *  necessary without further intervention from you. This means you will
icculus@10262
   546
 *  eventually run out of memory if you aren't routinely dequeueing data.
icculus@10262
   547
 *
icculus@10262
   548
 *  Capture devices will not queue data when paused; if you are expecting
icculus@10262
   549
 *  to not need captured audio for some length of time, use
icculus@10262
   550
 *  SDL_PauseAudioDevice() to stop the capture device from queueing more
icculus@10262
   551
 *  data. This can be useful during, say, level loading times. When
icculus@10262
   552
 *  unpaused, capture devices will start queueing data from that point,
icculus@10262
   553
 *  having flushed any capturable data available while paused.
icculus@10262
   554
 *
icculus@10262
   555
 *  This function is thread-safe, but dequeueing from the same device from
icculus@10262
   556
 *  two threads at once does not promise which thread will dequeued data
icculus@10262
   557
 *  first.
icculus@10262
   558
 *
icculus@10262
   559
 *  You may not dequeue audio from a device that is using an
icculus@10262
   560
 *  application-supplied callback; doing so returns an error. You have to use
icculus@10262
   561
 *  the audio callback, or dequeue audio with this function, but not both.
icculus@10262
   562
 *
icculus@10262
   563
 *  You should not call SDL_LockAudio() on the device before queueing; SDL
icculus@10262
   564
 *  handles locking internally for this function.
icculus@10262
   565
 *
icculus@10262
   566
 *  \param dev The device ID from which we will dequeue audio.
icculus@10262
   567
 *  \param data A pointer into where audio data should be copied.
icculus@10262
   568
 *  \param len The number of bytes (not samples!) to which (data) points.
icculus@10262
   569
 *  \return number of bytes dequeued, which could be less than requested.
icculus@10262
   570
 *
icculus@10262
   571
 *  \sa SDL_GetQueuedAudioSize
icculus@10262
   572
 *  \sa SDL_ClearQueuedAudio
icculus@10262
   573
 */
icculus@10262
   574
extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
icculus@10262
   575
icculus@10262
   576
/**
icculus@9012
   577
 *  Get the number of bytes of still-queued audio.
icculus@9012
   578
 *
icculus@10262
   579
 *  For playback device:
icculus@9012
   580
 *
icculus@10262
   581
 *    This is the number of bytes that have been queued for playback with
icculus@10262
   582
 *    SDL_QueueAudio(), but have not yet been sent to the hardware. This
icculus@10262
   583
 *    number may shrink at any time, so this only informs of pending data.
icculus@10262
   584
 *
icculus@10262
   585
 *    Once we've sent it to the hardware, this function can not decide the
icculus@10262
   586
 *    exact byte boundary of what has been played. It's possible that we just
icculus@10262
   587
 *    gave the hardware several kilobytes right before you called this
icculus@10262
   588
 *    function, but it hasn't played any of it yet, or maybe half of it, etc.
icculus@10262
   589
 *
icculus@10262
   590
 *  For capture devices:
icculus@10262
   591
 *
icculus@10262
   592
 *    This is the number of bytes that have been captured by the device and
icculus@10262
   593
 *    are waiting for you to dequeue. This number may grow at any time, so
icculus@10262
   594
 *    this only informs of the lower-bound of available data.
icculus@9012
   595
 *
icculus@9012
   596
 *  You may not queue audio on a device that is using an application-supplied
icculus@9012
   597
 *  callback; calling this function on such a device always returns 0.
icculus@10262
   598
 *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
icculus@10262
   599
 *  the audio callback, but not both.
icculus@9012
   600
 *
icculus@9012
   601
 *  You should not call SDL_LockAudio() on the device before querying; SDL
icculus@9012
   602
 *  handles locking internally for this function.
icculus@9012
   603
 *
icculus@9012
   604
 *  \param dev The device ID of which we will query queued audio size.
icculus@9012
   605
 *  \return Number of bytes (not samples!) of queued audio.
icculus@9012
   606
 *
icculus@9012
   607
 *  \sa SDL_QueueAudio
icculus@9012
   608
 *  \sa SDL_ClearQueuedAudio
icculus@9012
   609
 */
icculus@9012
   610
extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
icculus@9012
   611
icculus@9012
   612
/**
icculus@10262
   613
 *  Drop any queued audio data. For playback devices, this is any queued data
icculus@10262
   614
 *  still waiting to be submitted to the hardware. For capture devices, this
icculus@10262
   615
 *  is any data that was queued by the device that hasn't yet been dequeued by
icculus@10262
   616
 *  the application.
icculus@9012
   617
 *
icculus@10262
   618
 *  Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
icculus@10262
   619
 *  playback devices, the hardware will start playing silence if more audio
icculus@10262
   620
 *  isn't queued. Unpaused capture devices will start filling the queue again
icculus@10262
   621
 *  as soon as they have more data available (which, depending on the state
icculus@10262
   622
 *  of the hardware and the thread, could be before this function call
icculus@10262
   623
 *  returns!).
icculus@9012
   624
 *
icculus@9012
   625
 *  This will not prevent playback of queued audio that's already been sent
icculus@9012
   626
 *  to the hardware, as we can not undo that, so expect there to be some
icculus@9012
   627
 *  fraction of a second of audio that might still be heard. This can be
icculus@9012
   628
 *  useful if you want to, say, drop any pending music during a level change
icculus@9012
   629
 *  in your game.
icculus@9012
   630
 *
icculus@9012
   631
 *  You may not queue audio on a device that is using an application-supplied
icculus@9012
   632
 *  callback; calling this function on such a device is always a no-op.
icculus@10262
   633
 *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
icculus@10262
   634
 *  the audio callback, but not both.
icculus@9012
   635
 *
icculus@9012
   636
 *  You should not call SDL_LockAudio() on the device before clearing the
icculus@9012
   637
 *  queue; SDL handles locking internally for this function.
icculus@9012
   638
 *
icculus@9012
   639
 *  This function always succeeds and thus returns void.
icculus@9012
   640
 *
icculus@9012
   641
 *  \param dev The device ID of which to clear the audio queue.
icculus@9012
   642
 *
icculus@9012
   643
 *  \sa SDL_QueueAudio
icculus@9012
   644
 *  \sa SDL_GetQueuedAudioSize
icculus@9012
   645
 */
icculus@9012
   646
extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
icculus@9012
   647
icculus@9012
   648
icculus@9012
   649
/**
slouken@3407
   650
 *  \name Audio lock functions
slouken@7191
   651
 *
slouken@3407
   652
 *  The lock manipulated by these functions protects the callback function.
slouken@7191
   653
 *  During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
slouken@3407
   654
 *  the callback function is not running.  Do not call these from the callback
slouken@3407
   655
 *  function or you will cause deadlock.
slouken@0
   656
 */
gabomdq@7678
   657
/* @{ */
slouken@337
   658
extern DECLSPEC void SDLCALL SDL_LockAudio(void);
icculus@1964
   659
extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
slouken@337
   660
extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
icculus@1964
   661
extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
gabomdq@7678
   662
/* @} *//* Audio lock functions */
slouken@0
   663
slouken@3407
   664
/**
slouken@3407
   665
 *  This function shuts down audio processing and closes the audio device.
slouken@0
   666
 */
slouken@337
   667
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
icculus@1964
   668
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
icculus@1964
   669
slouken@0
   670
/* Ends C function definitions when using C++ */
slouken@0
   671
#ifdef __cplusplus
slouken@0
   672
}
slouken@0
   673
#endif
slouken@0
   674
#include "close_code.h"
slouken@0
   675
slouken@10638
   676
#endif /* SDL_audio_h_ */
slouken@1895
   677
slouken@1895
   678
/* vi: set ts=4 sw=4 expandtab: */